Voice over IP (VoIP)
Fundamentals
112
9.2.5 Call Set-up
Various IP-based protocols are used for system telephony via the Internet protocol
(“IP”) (see also
starting on page 129). Multiple TCP connections are
made between an IP telephone and OpenCom 100 for the telephone’s start pro-
cedure, registration and signalling. All voice data are directly exchanged between
IP telephones using the RTP (“Realtime Transport Protocol”) protocol.
Channels on a
are allocated for making a telephone con-
nection with an ordinary terminal or for dial tones. The MGW converts IP voice
data into PCM data streams used with conventional telephony and vice versa. For
this, IP voice data are exchanged between the IP telephone and the gateway.
Tip:
Switching between voice data channels may cause a slight
delay in some circumstances. For example: when accepting a
call on an IP telephone, headset users should wait about one
second before answering.
9.2.6 Useful Services
The type of data compression used for VoIP prevents these types of connections
from using certain services. Take these notes into account especially if you want to
use connections made via Q.SIG-IP or SIP:
■
ISDN data services can not be used
■
Faxes can only be sent using the uncompressed G.711 codec
■
DTMF dial tones are only received by the other party if the uncompressed
G.711 codec is used. Alternatively, DTMF dial tones can be transferred using the
Internet standard RFC 2833/4733. For this, the “DTMF (RFC4733)” codec needs
to be activated for the codec configuration on the
Telephony
:
Extended
:
VoIP
profile
page.
■
Analogue modems can not be used
Tip:
Configure the actual usage for the a/b ports, e.g. set them to
Fax
or
Data (analogue)
. Connections from and to these a/b
ports will then be made using uncompressed or ISDN connec-
tions where possible.
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